In telephone conferencing a number of voice signals from each telephone set involved in the conference must be combined and transmitted back to each of the telephone sets. The number of voice signals can be combined at a central exchange for example. With conventional analog telephones, the combination of the signals is straightforward. The several signals need only be summed together. To avoid overdrive of the final signal, each summed signal is generally reduced by an amount proportional to the number of signals being combined. For example, in a three-way conference, one-third of each signal is summed.
Conventional analog telephones are being replaced by digital telephones. In such systems, the analog signals are sampled at each telephone set at a rate of twice the band width of the analog signal or about eight kilohertz. Each sample is then quantized as one of a discrete set of prechosen values and encoded as a digital word which is then transmitted over the telephone lines. With eight bit digital words, for example, the analog sample is quantized to 2.sup.8 or 256 levels, each of which is designated by a different eight bit word. In linear pulse code modulation (PCM) systems, the 256 possible values of the digital word are linearly related to corresponding analog amplitudes. As a result, the digital values can be summed as with the combination of analog signals.
Because most speech is found at the lower analog signal amplitudes, encoding techniques have been developed which maintain high resolution at the lower amplitudes but which provide lesser resolution at higher amplitudes. Such approaches reduce the number of bits required in each word. An example of such an encoding technique is the .mu. law technique by which the quantization levels are based on a logarithmic function. As a result of the nonlinear relationship between the encoded digital signals and the underlying analog voice signals, the digital signals cannot simply be proportionately reduced and summed. Such direct summation results in severe distortion. Typically, as shown in FIG. 1, such .mu. law encoded signals which must be combined for teleconferencing are first converted back to analog signals through an inverse .mu. law device 12 and the analog signals are summed in summer 14. The resultant combined signals are then again encoded through a .mu. law encoder 16. This approach introduces extra quantization noise and more hardware implementation due to the extra conversion from digital to analog and back to digital.
An alternative to reverting back to analog form is to determine an "Active Speaker" by comparing each conferee's digital samples during each of sequential time frames. Once the active speaker is determined, only the digital voice signal from that speaker is transmitted to the others through a time frame. Such an approach can result in chopping of words as the active speaker status is passed from one conferee to another and does prevent simultaneous transmission from multiple speakers.